incoming called-number is not working with sip, but destination-pattern is on CME

[mercdecember]

I have setup a CME lab environment with voip.ms.  I am able to make calls in and out with no issues.  However my incoming dial-peer is only working if I have "Destination-pattern .T" setup on it, if I replace it with "incoming called-number .%", then I cannot receive calls.  Any Idea why I cannot use incoming called-number?  Below is my conifg

voice translation-rule 1
 rule 1 /^9/ //
!
voice translation-rule 2
 rule 1 /^.*/ /91XXXX1462/
!
voice translation-rule 3
 rule 1 /^91XXXX1\(...\)$/ /\1/
!
!
voice translation-profile INCOMING_CALLS
 translate called 3
!
voice translation-profile OUTGOING_CALLS
 translate calling 2
 translate called 1

dial-peer voice 7 voip
 translation-profile outgoing OUTGOING_CALLS
 destination-pattern 9[2-9]......
 session protocol sipv2
 session target dns:newyork4.voip.ms
 no voice-class sip localhost
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 11 voip
 translation-profile outgoing OUTGOING_CALLS
 destination-pattern 91[2-9].........
 session protocol sipv2
 session target dns:newyork4.voip.ms
 no voice-class sip localhost
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 1 voip
 translation-profile incoming INCOMING_CALLS
 destination-pattern .T
 session protocol sipv2
 session target dns:newyork4.voip.ms
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

[Ayodeji Okanlawon]

Have you tried only "incoming called number." without the %. I have personally never used incoming-called number.%, just incoming-called number. works for me

[mercdecember]

Tried it, it did not work.

[Ayodeji Okanlawon]

Please configure the ff:

dial-peer voice 1 voip
 translation-profile incoming INCOMING_CALLS
 no destination-pattern .T

incoming called number.
 session protocol sipv2
 no session target dns:newyork4.voip.ms
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

Then enable the ff debugs and do a test call..

debug voip ccapi inout

debug ccsip messages..

Attach logs here include calling and called number

[mercdecember]

no debug voip ccapi inout output

but here is the debug ccsip messages

[Ayodeji Okanlawon]

From the logs this INVITE is sent to an invalid ip address..74.101.112.85--

Which IP is this? It looks like this is not configured on your ccme..

Received:
INVITE sip:9142281462@74.101.112.85 SIP/2.0

Via: SIP/2.0/UDP 107.6.67.238:5060;branch=z9hG4bK6efa0445;rport

Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'

Via: SIP/2.0/UDP 107.6.67.238:5060;branch=z9hG4bK6efa0445;rport

From: "9144410197" <sip:9144410197@107.6.67.238>;tag=as0257aead

[mercdecember]

that is my wan ip, its on dhcp so it changes

[Ayodeji Okanlawon]

This is where your issue is. This will not work like this. You will need to send the call to the IP address of your ccme server. I am surprised you are even receiving calls at all. I would like to see the debugs for a working call with destination-pattern .T

[mercdecember]

how would I do that:

this is my sip-ua profile

sip-ua
 credentials username XXXXXXX password 7 <removed> realm newyork4.voip.ms
 authentication username XXXXXXX password 7 <removed> realm newyork4.voip.ms
 no remote-party-id
 retry invite 3
 retry register 3
 timers register 150
 registrar 1 dns:newyork4.voip.ms expires 200
 sip-server dns:newyork4.voip.ms
!

[Ayodeji Okanlawon]

How is your WAN IP on dhcp? Why is it on DHCP..It should never be. What does your WAN gateway connects to? is this topology correct?

Can you describe your infrastructure here.. How does your ccme connect to the ITSP. Please include all the devices in the mix. On another note I think you should get your data team involved into routing calls from your ITSP to your internal network..

[mercdecember]

It is becuase this is a lab in my apartment with a home ISP.  It is connected to voip.ms

sip-ua
 credentials username XXXXXXX password 7 <removed> realm newyork4.voip.ms
 authentication username XXXXXXX password 7 <removed> realm newyork4.voip.ms
 no remote-party-id
 retry invite 3
 retry register 3
 timers register 150
 registrar 1 dns:newyork4.voip.ms expires 200
 sip-server dns:newyork4.voip.ms

[mercdecember]

Also here are the devices

The 2811 router as my SIP VOIP Gateway, and then it is connected via trunk to a Cisco 892FSP which is just used as an internet facing router.

[Ayodeji Okanlawon]

In this scenario you will need to do the ff

1. Create a public ip for your ccme. This will be routed through your wan ip. All traffic to this ip must go via your wan gateway

2. on the wan gateway you will need to configure NAT, to translate this public IP to the internal IP of your ccme.

3. Your wan gateway must be able to do SIP ALG, so that sip messages are properly modified during the NAT conversion.

[mercdecember]

I am not sure you guys are understanding this, this is in a lab environment.  My router can get to the outside, I am using a home ISP so there is no static ip's.  I can make calls with no issues, I can receive calls with no issues.  All I need to know is why when I receive calls neither the answer-address nor the incoming called-number command work.  If I use destination-pattern .T then all of a sudden I can receive calls.

[Ayodeji Okanlawon]

okay, can you please send the following logs for a working call with destination-pattern.T ( I did asked for this earlier)

debug voip ccapi inout

debug ccsip messages